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BT & BTIP
Genesys PureCloud on Premises
Orange SA au capital de 10 595 541 532 €
78 rue Olivier de Serres 75505 Paris Cedex 15
380 129 866 RDC Paris
1 of 17
Business Talk & Business Talk IP For Genesys PureCloud Edge
on Premises
Version addressed in this guide : 1.0.07616
Information included in this document is dedicated to customer equipment (IPBX, TOIP ecosystems) connection to Business Talk IP service : it shall not be used for other goals or in another context.
Document Version
Version of 26/03/2019
BT & BTIP
Genesys PureCloud on Premises
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78 rue Olivier de Serres 75505 Paris Cedex 15
380 129 866 RDC Paris
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Table of Contents
Table of Contents ...................................................................................................................... 2
Goal of this document ............................................................................................................... 3
1 ARCHITECTURE OVERVIEW .............................................................................................. 4 1.1 Architecture ............................................................................................................................... 4
2 PARAMETERS to be PROVIDED by CUSTOMER to ACCESS BT/BTIP service .................... 6
3 Business Talk & BTIP certified versions ............................................................................... 7 Please refer to ‘https://help.mypurecloud.com/edge-media-tier-release-notes/’ web page for more
details about the versions supported by Genesys. ...................................................................... 7 3.1 Genesys PureCloud Edge .......................................................................................................... 7 3.2 Endpoints and applications ........................................................................................................ 7
4 SIP TRUNKING CONFIGURATION CHECKLIST .................................................................. 8 4.1 External Trunk ............................................................................................................................ 8 4.2 Phone Trunk ............................................................................................................................ 11 4.3 Edges ...................................................................................................................................... 13 4.4 Call routing ............................................................................................................................... 14 4.5 N+1 Redundancy ..................................................................................................................... 15 4.6 Call Admission Control ............................................................................................................. 15
5 SIP SOFTPHONE CONFIGURATION WITH EDGE ............................................................. 16 5.1 Base Phone Settings ................................................................................................................ 16 5.2 Phone ...................................................................................................................................... 17
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Goal of this document
The aim of this document is to list technical requirements to ensure the interoperability between Genesys
PureCloud Edge solution on Customer Premises with OBS Business Talk & Business Talk IP SIP
services, here after so-called “service”.
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1 ARCHITECTURE OVERVIEW
1.1 Architecture
Notes:
In the diagram above, the SIP and proprietary internal flows are hidden.
In this architecture
- All ‘SIP trunking’ signalling flows are carried by the PureCloud Edge and routed on the main BVPN
connection.
- PureCloud Edge anchors systematically the RTP flow for calls to/from Orange a-SBC. Therefore, the
RTP flow is not direct between SIP phones/softphones and Orange a-SBC. But IP routing differs
from one site to another:
o For the Head Quarter site, media flows are just routed on the main BVPN connection.
o For Remote sites on BVPN, media flows are just routed on the local BVPN connection (=
distributed architecture).
o For Remote sites on Third Party WAN, media flows are routed through the Head Quarter
(through the Edge) and use the main BVPN connection (= centralized architecture, cf sizing
below).
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- Call scenario nb of voice channels/media resources used
EDGE WAN router* BTIP/BT
1 offnet call from/to the headquarter (HQ) 1 1 in HQ 1 in HQ
1 offnet call from/to a remote site (RS) on BVPN 2 2 in HQ
1 in RS
0 in HQ
1 in RS
1 offnet call from/to a remote site (RS) on TP Wan 2 1 in HQ BVPN
1 in HQ TP Wan 1 in RS TP Wan
0 in HQ 1 in RS
1 offnet call from/to a remote site which put the caller on
hold
2 3 in HQ (ou 2)
1 in RS 0 in HQ 1 in RS
1 offnet call from/to a remote site after transfer/forward to BTIP
2 2 in HQ 0 in RS
0 in HQ 2 in RS
1 forced onnet call from Headquarter to a remote site (= through Business Talk IP infrastructure)
2 3 in HQ 1 in RS
0 in HQ 0 in RS
*On the WAN router, 1 voice channel= 80Kb/s
Redundancy consideration
Edge appliances can be deployed in an N+1 configuration in order to provide local redundancy.
This requires that :
o All the Edges are on the same geographical site or distributed in multi sites with very
low latency between each site.
o All the Edges belong to the same PureCloud Edge Group.
The Orange a-SBC distributes the calls in round-robin on all the Edge IP addresses of the BTIP/BT
Sites.
Sizing consideration
Specific sizing approach has to be considered with PureCloud Edge due to the fact that the Edge
anchors systematically the RTP flow for calls to/from Orange a-SBC. Therefore, the RTP flow is not
direct between SIP phones/Softphones and Orange a-SBC.
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2 PARAMETERS to be PROVIDED by CUSTOMER to ACCESS BT/BTIP service
IP addresses marked in red have to be indicated by the Customer, depending on Customer
architecture scenario
N+1 Redundancy architecture - HQ Site architecture
Level of Service
Customer IP addresses used by the service
Nominal Backup
N+1 redundancy: All edges are member of the same Edge group. The incoming load sharing is done using a-SBC round robin to all Edges. If an Edge passes out of Service, the secondary Edge becomes the new Primary for each softphone which registers on it, even when the previous Edge comes back In Service. If existing, a 3rd Edge becomes Secondary. Softphones or SIP phones are hosted on a physical site with or without Edge. A site from BT/BTIP point of view corresponds to a physical site with or without phones/softphones.
- Local Edge appliance redundancy - User registration redundancy: User is registered on 2 edges. One edge is seen as Primary for the user, the other one is seen as Secondary.
EDGE 1 IP@ EDGE 2 IP@ EDGE 3 IP@
….
NA
Remote Site (RS) architecture Level of Service
Customer IP addresses used by the service
Nominal Backup
Remote site No survivability, no trunk redundancy.
NA NA
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3 Business Talk & BTIP certified versions
Please refer to ‘https://help.mypurecloud.com/edge-media-tier-release-notes/’ web page for more
details about the versions supported by Genesys.
3.1 Genesys PureCloud Edge
GENESYS PURECLOUD EDGE – software versions
Reference product Software version
Certification
: Certified
NS : No supported
Certified "Loads" Restrictions
Genesys PureCloud Edge
1.0.0 1.0.07616
3.2 Endpoints and applications
Endpoints and applications
Reference product Software version
NA: not applicable
Certification
: Certified
NS : No supported
Genesys PureCloud
Edge version Comments
Genesys endpoints
PureCloud Softphone 2016 R2
micro programme : 16.2.0.1888
1.0.07616
Audiocodes phones
Audiocodes 400HD - Not tested -
PureCloud supports a number of types of phones. Refer to the information provided by Genesys to find details about the phones that PureCloud supports.
Polycom phones
Polycom SoundPoint IP - Not tested -
Polycom RealPresence Trio
- Not tested -
Polycom VVX - Not tested -
Polycom SoundStation IP
- Not tested -
Spectralink phones
Spectralink 8000 Portfolio Handsets
- Not tested -
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4 SIP TRUNKING CONFIGURATION CHECKLIST
The checklist below presents all the steps of configuration required for interoperability between BTIP/BT
and Genesys PureCloud Edge.
Two external trunks must be created, one to each Orange a-SBC. Each “external trunk” instance applies
to all edges.
Each edge uses these 2 trunks in Primary/Backup mode (see below). Each trunk is used for inbound and
outbound calls.
One “Phone trunk” must be created for SIP Phone (Purecloud softphones), it applies to all edges.
4.1 External Trunk
Steps Parameter Name Value (example)
Creation et Configuration for an external trunk to a-SBC
Step 1 TelephonyTrunksExternal Trunks « Trunk Name »
General
Parameter Name
1 External Trunk Name
2 Type
3 Trunk State
4 Protocol
5 Listen Port
Value
1 Trunk Name (ex : SBC_113) 2 External SIP
3 In-Service
4 UDP
5 5060
Outbound
Parameter Name
1 SIP Servers or Proxies
2 Digest Authentication
3 Calling Address
Value
1 SBC @IP:port (Ex : 172.22.246.33:5060)
2 Disable
3 Phone Number of Call Center (Ex : +0800PQMCDU)
Availability
Parameter Name
1 State
2 Interval
Value
1 Enable
2 600 sec
Registration
Parameter Name
1 State
Value
1 Disable
Sip Access Control
Parameter Name
1 Use Source Address
2 Allow All
3 Allow the Following Addresses
Value
1 Yes
2 uncheck
3 a-SBC @IP
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External Trunk Configuration/General
Parameter Name
1 Call Draining
2 Language
3 Max Calls
4 Max Call Rate
5 Max Dial Timeout
6 Asserted Identity
Value
1 Enabled 2 English
3 350
4 40/5s 5 120 sec 6 Disabled
External Trunk Configuration/Transport
Parameter Name
1 Transport DSCP Value
2 Retryable Reason Codes
3 Retryable Cause Codes
4 UDP T1 timeout
5 UDP T2 timeout
6 UDP Max Packet retry
7 UDP Max Invite Retry
Value
1 18 (24.011000)CS3
2 500-599
3 1-5,25,27,28,31,34,38,41,42,
44,46,62,63,79,91,96,97,99,
100,103 4 500ms 5 1s 6 4 7 3
External Trunk Configuration/Identity
Parameter Name Inbound
1 Identity Type
Outbound
2 Apply header privacy
3 Apply User privacy
4 Calling/Address Transformation
5 Calling/Address Digits length
6 Calling/Address Omit+Prefix
7 Called/Address Transformation
8 Called/Address Digits length
9 Called/Address Omit+Prefix
Value
1 From
2 Disabled or Enabled
3 Disabled 4 5 0
6 Disabled 7 8 0
9 Disabled
External Trunk Configuration/Media
Parameter Name
1 DSCP Value
2 Media Method
3 Preferred Codec List
4 SRTP Cipher Suite List
5 Ringback
6 Disconnect on Idle RTP
DTMF Setting
7 DTMF Payload
8 DTMF Method
Recording
9 Line Recording
10 Dual Channel
11 Continue on External Transfert
12 Audio Format
13 Automatic Level Control
14 Recording/ Consent Required
Value
1 2E (46,101110) EF
2 Normal
3 Codec (Audio/PCMA, or Audio/G729)
4 AES_CM_128_HMAC_SHA1_80 5 Enabled 6 Enabled
7 101 8 RTP Event
9 Enabled
10 Enabled
11 Enabled 12 audio/ PCMA
13 Disabled
14 Disabled
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External Trunk Configuration/Protocol
Parameter Name
Header/Invite
1 Conversation Headers
2 From Header Hostname
3 Routing Address
4 Diversion Method
5 Asserted Identity Header
6 Max Diversion Entries
7 Request Target Address
User to User Information (UUI)
8 UUI Passthrough
9 Header/Type
10 Encoding Format
11 Protocol Discriminator
12 Static User Data
13 Header/name
14 Value
15 Priority
Take Back and Transfer
16 Enable Take Back and transfer
Value
1 Enabled
2 Automatically generate from Edge
network Interface
3 Request-URI 4 None 5 P-Asserted-Identity
6 4 7
8 Enabled
9 User-to-User
10 Hex
11 04
12 Disable or Enabled
13 User-to-User
14 Ex : 0468656c6c6f20776f726c64;encoding=hex;purpose=isdn-interwork;content=isdn-uui
15 Low
16 Disabled
External Trunk Configuration/Diagnostic
Parameter Name
1 Media Capture
2 protocol Capture
Value
17 Disabled
18 Disabled
External Trunk Configuration/Custom
Parameter Name Value
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4.2 Phone Trunk
Steps
Parameter Name
Value (example)
Creation and Configuration for Phone trunk
Step 1 TelephonyTrunksPhone Trunks « Trunk Name »
General
Parameter Name
1 Phone Trunk Name
2 Type
3 Trunk State
4 Protocol
5 Listen Port
Value
1 Phone Trunk Name (to_softphone)
2 SIP
3 In-Service (check) 4 UDP
5 8060
Registration
Parameter Name
1 Max Registration Rate
Value
1 600/1m
Sip Access Control
Parameter Name
1 Use Source Address
2 Allow All
Value
1 No
2 Check
Connection Configuration /General
Parameter Name
1 Language
Calls
2 Max Calls
3 Max Call Rate
4 Max Dial Timeout
Value
1 English – United States
2 100
3 40 4 60 sec
Connection Configuration/Transport
Parameter Name
1 Transport DSCP Value
2 Retryable Reason Codes
3 Retryable Cause Codes
UDP Settings
4 UDP T1 timeout
5 UDP T2 timeout
6 UDP Max Packet retry
7 UDP Max Invite Retry
Value
1 18 (24.011000)CS3
2 500-599
3 1-5,25,27,28,31,34,38,41,42,
44,46,62,63,79,91,96,97,99,
100,103 4 500ms 5 1s 6 4 7 3
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Connection Configuration /Media
Parameter Name
1 DSCP Value
2 Media Method
3 Preferred Codec List
4 SRTP Cipher Suite List
5 Disconnect On Idle RTP
DTMF Setting
1 DTMF Payload
2 DTMF Method
Value
1 2E (46,101110) EF
2 Normal
3 Codecs (Audio/PCMA or AUDIO/g729)
4 AES_CM_128_HMAC_SHA1_80 5 Enable
6 101
7 RTP Event
Connection Configuration /Protocol
Parameter Name
Header/Invite
1 Conversation Headers
Value
8 Enabled
Connection Configuration /Authentification
Parameter Name
1 Inbound Digest
Authentication
2 Authorization Methods
3 Realm
4 User Name
5 Password
Value
1 Disabled
2 REGISTER
3 ININ
4 5
Connection Configuration /Provisioning
Parameter Name
1 Require Authentications
2 Prefer http Redirects
3 Require HTTPS
4 External Provision Server
5 HTTP Port
6 HTTPS Port
Value
1 Disabled
2 Disabled
3 Disabled
4
5 8098
6 8099
Connection Configuration /Diagnostic
Parameter Name
1 Media Capture
2 protocol Capture
Value
1 Disabled
2 Disabled
Connection Configuration /Custom
Parameter Name Value
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4.3 Edges
Steps
Parameter Name
Value (example)
Site Configuration
Step 1
TelephonyEdges« Edge Name »
General
Parameter Name
1 Edge Name
2 Site
3 Edge Group
4 Network Interface for Internal
Edge Communication
Value
1 Edge Name 2 Site
3 Edge Group Name 4 NIC name (ex: Port LAN 1)
Network InterfacesInterface Name External Trunks
Parameter Name
1 Use the following trunks
2 select External Trunks
1 Check
2 Trunk to Primary a-SBC
Trunk to Secondary a-SBC
Network InterfacesInterface Name Phone Trunks
Parameter Name
1 Use the following trunks
2 select Phone Trunks
1 Check
2 Phone Trunk Name
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4.4 Call routing
Call routing for Primary a-SBC and Backup a-SBC is set at the “site level”
Steps
Parameter Name
Value (example)
Site Configuration
Step 1
TelephonySites Outbound Routes « Outbound Route Name »
General
Parameter Name
1 Outbound Route Name
2 Description
3 State
4 Classifications
Value
1 Default Outbound Route
2 Description
3 Enabled
4 Emergency, National,
International
distribution Pattern
Parameter Name
1 Sequential
2 Random
1 Check
2 Uncheck
External Trunks
Primary a-SBC Trunk Name (at the top)
Secondary a-SBC Trunk Name (at the bottom)
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4.5 N+1 Redundancy
Steps
Parameter Name
Value (example)
Site Configuration
Step 1 TelephonySites « Site Name »
General
Parameter Name
1 Site Name
2 Description
3 Location
Value
1 Site Name 2 Description
3 Location
Phone Edge Assignements
Parameter Name
1 Use these Sites
2 Primary Sites
3 Secondary Sites
1 Check
2 sites list (all sites containing
Edges)
3 sites list (all sites containing
Edges)
4.6 Call Admission Control
The CAC is managed by the "Max Calls" options in the trunk configuration. The same Pure Cloud
trunk is used from one a-SBC to all edges, so this value is the global CAC, in and out, for all the
edges.
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5 SIP SOFTPHONE CONFIGURATION WITH EDGE
This chapter describes the Purecloud configuration for Purecloud SIP phones. All options are listed.
Two configurations menus are used :
- “Based settings” for global configuration for a phone type.
- “Phone” to create an instance of a phone type and optionally to adjust the parameters set in the base
settings.
5.1 Base Phone Settings
Steps
Parameter Name
Value (example)
Base Phone settings
Step 1 TelephonyPhone ManagementBase Settings “Base Settings Name”
General
Parameter Name
1 Base Settings Name
2 Phone Make and Model
Value
1 "BaseName" 2 PurecloudSoftphone
Phone Configuration/General
Parameter Name
1 Firmware Version
2 Dynamic Reload
Value
1 "the last version"
2 Enable
Phone Configuration/Media
Parameter Name
1 DSCP Value
2 RTP Audio Port Start Range
3 Preferred Codec List
DTMF Setting
1 DTMF Payload
2 DTMF Method
Value
1 2E (46,101110) EF
2 16384
3 Codec (Audio/PCMA or Audio G729)
1 101
2 RTP Event
Phone Configuration/Network
Parameter Name
1 Provision
2 TLS Authority ID
Signaling
1 DSCP
Value
1 From Edges within the Site
2
1 18 (24. 011000)CS3
Phone Configuration/Custom
Parameter Name Value
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5.2 Phone
Steps
Parameter Name
Value (example)
Phone création and configuration
Step 1 TelephonyPhone ManagementPhones"Phone Name"
General
Parameter Name
1 phone Name
2 Base settings
3 Site
4 Hardware ID
Value
1 "PhoneName" 2 "Base_Setting_Name"
3 "site name" 4 "station FQDN" where the
softphone is installed
Phone Configuration/General
Parameter Name
1 Firmware Version
2 Dynamic Reload
Value
1 "the last version" 2 Enable
Phone Configuration/Media
Parameter Name
1 DSCP Value
2 RTP Audio Port Start Range
3 Preferred Codec List
DTMF Setting
1 DTMF Payload
2 DTMF Method
Value
1 2E (46,101110) EF
2 16384
3 Codec (Audio/PCMA, Audio G729)
1 101 2 RTP Event
hone Configuration/Network
Parameter Name
1 Provision
2 TLS Authority ID
Signaling
3 DSCP
Value
1 From Edges within the Site
2
3 18 (24. 011000)CS3
Phone Configuration/Custom
Parameter Name Value